1. Field of the Invention
The present invention relates to a method and an apparatus for providing a voice service through a packet network in a mobile communication system. More particularly, the present invention relates to a method and an apparatus for efficiently setting the size of packet data according to a voice service.
2. Description of the Related Art
Recently, mobile communication systems are developing high speed and high quality wireless data packet communication systems in order to provide data services and multimedia services beyond initial voice-based services. The universal mobile telecommunication service (UMTS) system (which is the 3rd mobile communication system) employing a wideband code division multiple access (W-CDMA) scheme based on the global system for mobile communications (GSM) and the general packet radio services (GPRS), which are European Mobile Communication Systems, consistently provides a service allowing users employing mobile phones or computers to transmit packet-based text or digitalized voice data, video and multimedia data at a high speed of at least 2 Mbps, regardless of the locations of the users.
This UMTS system employs an access concept such as a packet switching scheme based on an packet protocol including Internet Protocol (IP). The third generation partnership project (3GPP) performing standardization work for such a UMTS communication system has further proposed a plan for supporting Voice over Internet protocol (VoIP) communication.
The VoIP denotes a communication scheme for making a voice frame generated from a voice codec into an Internet Protocol/User Datagram Protocol/Real-time Transport Protocol (IP/UDP/RTP) packet to be transmitted. If the VoIP is used, a voice service may be provided through the packet network.
FIG. 1 schematically illustrates the structure of a wireless access network (UTRAN) of a typical asynchronous mobile communication system (UMTS). The mobile communication system includes a core network (CN) 100 and a plurality of radio network subsystems (RNSs) 110 and 120. The RNSs 110 and 120 construct an UMTS Terrestrial Radio Access Network (UTRAN). The CN 100 includes a serving GPRS supporting node (SGSN) (not shown) and a gateway GPRS support node (GGSN) (not shown) in order to connect the UTRAN to a packet data network such as the Internet (not shown).
The RNSs 110 and 120 include radio network controllers (RNCs) 111 and 112, and a plurality of Node Bs 115, 113, 114 and 116. In detail, the RNS 110 includes the RNC 111 and the Node Bs 115 and 113, and the RNS 120 includes the RNC 112 and the Node Bs 114 and 116. The RNCs 111 and 112 are divided into a serving RNC, a drift RNC and a control RNC according to the roles of the RNCs. The serving RNC manages information of each UE and provides data transmission together with the CN. The drift RNC directly makes wireless connection with a UE. The control RNC controls wireless resources of each Node B.
The RNCs 111 and 112 are connected with the Node Bs 115, 113, 114 and 116 through an interface denoted as “lub”, and the connection between the RNC 111 and 112 is achieved through an interface denoted as “lur”. In addition, although not shown in FIG. 1, the UE 130 is connected with the UTRAN through an interface denoted as “Uu”.
The RNCs 111 and 112 allocate radio resources to the Node Bs 115 and 113 and Node Bs 114 and 116 managed by the RNCs 111 and 112, respectively, and the Node Bs 115, 113, 114 and 116 actually provide the UE, or terminal 130 with radio resources allocated from the RNCs 111 and 112. The radio resources are provided according to cells, and radio resources provided by each Node B are used for a specific cell managed by a corresponding Node B.
The terminal 130 establishes a radio channel by using radio resources for specific cells managed by the Node Bs 115, 113, 114 and 116, and makes data communication through the established radio channel. Since the terminal 130 recognizes only a physical channel formed according to cells, the discrimination between a cell and a Node B has little significance. Accordingly, in the following description, the cell and the Node B will be used without discrimination there between.
FIG. 2 illustrates the structure of the mobile communication system performing the VoIP communication.
A terminal 200 includes a codec 206 for converting voice into a voice frame, an IP/UDP/RTP protocol layer 205 for making the voice frame of the codec 206 into an IP/UDP/RTP packet, a packet data convergence protocol (PDCP) layer 204 for compressing a header of the IP/UDP/RTP packet, and a radio link control (RLC) layer 203 for converting the IP/UDP/RTP packet into a format suitable for transmission through a radio channel. The terminal 200 further includes a medium access control (MAC) module 202 for delivering a packet sent from the RLC layer to the physical channel through a suitable transport channel and delivering data sent from the physical layer through a transport channel to the RLC layer through a suitable logical channel, and a physical layer (PHY) 201 for exchanging packet data with the MAC layer and the transport channel in connection with the MAC layer and the transport channel, and transmitting the packet data to a receiver through a radio channel.
Voice packet data transmitted by the terminal 200 is delivered to the RNC 220 through the PHY layer 211 of the Node B 210.
In addition, the RNC 220 includes a MAC layer 222, an RLC layer 223 and a PDCP layer 224 similarly with the terminal 200 so as to convert the received data into an original IP/UDP/RTP packet to be transmitted to a core network (CN) 230. The IP/UDP/RTP packet is transmitted to a counterpart through an IP network 240. In a terminal of the counterpart, the voice packet data is controlled and delivered in a manner substantially opposite to the above-described order.
Hereinafter, the role of the RLC layer will be described in greater detail.
Generally, the RLC layer has modes classified into an unacknowledged mode (UM), an acknowledged mode (AM), and a transparent mode (TM) according to the operation of the RLC layer. The VoIP communication is performed in the UM of the RLC layer. In the following description, the operation of the UM will be given.
A UM of the RLC layer in a transmitter forms data having a size suitable for transmission through a radio channel by performing division, concatenation or padding with respect to an RLC service data unit (RLC SDU) delivered from the upper layer, inserts information about the division/concatenation/padding into the data, and inserts a serial number into the data so as to make an RLC protocol data unit (RLC PDU). The RLC PDU is then delivered to the lower layer.
Accordingly, to deliver from the transmitter, a UM of the RLC layer in a receiver analyzes the serial number and the division/concatenation/padding information of the RLC PDU delivered from the lower layer and forms the RLC PDU into an RLC SDU to be delivered to the upper layer.
For reference, the operation of a TM of the RLC layer is achieved by delivering an RLC SDU sent from the upper layer to the lower layer as it is, or by delivering an RLC PDU sent from the lower layer to the upper layer as it is.
As described above, voice data generated from the codec 206 of the terminal 200 is converted into a VoIP packet through the IP/UDP/RTP protocol layer 205. The VoIP packet has a header compressed through the PDCP layer 204 constructed for an uplink transmission, and is made into data having a size suitable for radio channel transmission through the RLC layer 203, then channel-coded in the MAC/PHY layers 201 and 202 and transmitted through a radio channel.
The RLC PDU (wherein the PDU processed in the physical layer is called a ‘Transport Block’) is channel-decoded in the physical layer 211 of the Node B 210 and then transmitted to the RNC 220.
The RNC 220 transforms the RLC PDUs into a VoIP packet to be transmitted to the CN 230. The CN 230 delivers the VoIP packet to a counterpart through the IP network 240 or the PSTN 250. Downlink data transmission is achieved in a manner substantially opposite to the above-described order.
In this case, in the VoIP communication system, users who communicate with each other must use the same codecs (e.g., the codecs 206 and 244). If communication between the UMTS terminal 200 and a PSTN user is achieved, a predetermined device performs the conversion of the codec 254 between the PSTN and the UMTS CN.
FIG. 3 illustrates the structure of the conventional VoIP mobile communication system using an adaptive multi-rate codec.
Terminals 305 and 330 having AMR codecs, generate IP packets including AMR payloads 310 to be delivered to counterparts of the terminals 305 and 330 through access networks 315 and 325 and a core network 320. The access networks 315 and 325 include an RNC (not shown) and a Node B (not shown), and the core network 320 may be a GPRS support node (GSN) (not shown).
The AMR codec has a notable characteristic in that the size of a payload can be variably adjusted according to environments of a radio link. For example, the AMR codec generating a small payload under an inferior link environment, generates a large payload if the link has a superior environment. Such an adjustment of the payload size refers to the change of a codec mode. The codec mode may be changed when the receiver directs a preferred codec mode to the transmitter by using a codec mode request (CMR) field of the header attached to the AMR payload 310.
For example, if the terminal 330 recognizes the necessity of changing the codec mode, the terminal 330 sets the CMR field of the header of the AMR payload 310 to a suitable value so as to transmit the AMR payload 310 to the terminal 305. Then, the terminal 305 adjusts the size of the payload by controlling the AMR codec to have the requested codec mode.
The AMR codec generates nine types of payloads including 56-bit payload, 112 bit-payload, 120-bit payload, 128-bit payload, 144-bit payload, 160-bit payload, 176-bit payload, 216-bit payload and 256-bit payload. The 56-bit payload among the payloads is used in a silent period; and the remaining eight payloads are each defined according to codec modes.
However, generally, the access networks 315 and 325 frequently fix the size of a packet to be used in a radio interval at a predetermined value. For example, the UTRAN frequently uses a 328-bit packet (also called a ‘transport block’) for a packet service.
Therefore, if the size of a packet is not changed, even though different sizes are given to an AMR payload of the packet according to the operation of the AMR codec, it is difficult to obtain a benefit according to an adaptive codec mode. That is, if the size of the packet is fixed at 328 bits even though the codec generates a 56-bit payload, an empty space in 328 bits is filled with padding bits and transmitted. Accordingly, the efficiency of radio resources may be degraded.
In addition, the 3rd generation mobile communication system has variously defined codec in addition to the AMR codec shown in FIG. 3, and these codecs make payloads having specific sizes.
That is, if the size of a packet used in the radio channel (or link) does not match the size of a payload generated from a specific codec, the conventional technique presents a problem of degrading the efficiency of radio resource utility.
Accordingly, a need exists for a system and method for efficiently processing a voice packet through a packet network in a mobile communication system.